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Diffstat (limited to 'media/libeffects/testlibs/AudioPeakingFilter.h')
| -rw-r--r-- | media/libeffects/testlibs/AudioPeakingFilter.h | 151 |
1 files changed, 151 insertions, 0 deletions
diff --git a/media/libeffects/testlibs/AudioPeakingFilter.h b/media/libeffects/testlibs/AudioPeakingFilter.h new file mode 100644 index 0000000..d0f49c9 --- /dev/null +++ b/media/libeffects/testlibs/AudioPeakingFilter.h @@ -0,0 +1,151 @@ +/* //device/include/server/AudioFlinger/AudioPeakingFilter.h +** +** Copyright 2009, The Android Open Source Project +** +** Licensed under the Apache License, Version 2.0 (the "License"); +** you may not use this file except in compliance with the License. +** You may obtain a copy of the License at +** +** http://www.apache.org/licenses/LICENSE-2.0 +** +** Unless required by applicable law or agreed to in writing, software +** distributed under the License is distributed on an "AS IS" BASIS, +** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. +** See the License for the specific language governing permissions and +** limitations under the License. +*/ + +#ifndef ANDROID_AUDIO_PEAKING_FILTER_H +#define ANDROID_AUDIO_PEAKING_FILTER_H + +#include "AudioBiquadFilter.h" +#include "AudioCoefInterpolator.h" + +namespace android { + +// A peaking audio filter, with unity skirt gain, and controllable peak +// frequency, gain and bandwidth. +// This filter is able to suppress introduce discontinuities and other artifacts +// in the output, even when changing parameters abruptly. +// Parameters can be set to any value - this class will make sure to clip them +// when they are out of supported range. +// +// Implementation notes: +// This class uses an underlying biquad filter whose parameters are determined +// using a linear interpolation from a coefficient table, using a +// AudioCoefInterpolator. +// All is left for this class to do is mapping between high-level parameters to +// fractional indices into the coefficient table. +class AudioPeakingFilter { +public: + // Constructor. Resets the filter (see reset()). + // nChannels Number of input/output channels (interlaced). + // sampleRate The input/output sample rate, in Hz. + AudioPeakingFilter(int nChannels, int sampleRate); + + // Reconfiguration of the filter. Changes input/output format, but does not + // alter current parameter values. Clears delay lines. + // nChannels Number of input/output channels (interlaced). + // sampleRate The input/output sample rate, in Hz. + void configure(int nChannels, int sampleRate); + + // Resets the filter parameters to the following values: + // frequency: 0 + // gain: 0 + // bandwidth: 1200 cents. + // It also disables the filter. Does not clear the delay lines. + void reset(); + + // Clears delay lines. Does not alter parameter values. + void clear() { mBiquad.clear(); } + + // Sets gain value. Actual change will only take place upon commit(). + // This value will be remembered even if the filter is in disabled() state. + // millibel Gain value in millibel (1/100 of decibel). + void setGain(int32_t millibel); + + // Gets the gain, in millibel, as set. + int32_t getGain() const { return mGain - 9600; } + + // Sets bandwidth value. Actual change will only take place upon commit(). + // This value will be remembered even if the filter is in disabled() state. + // cents Bandwidth value in cents (1/1200 octave). + void setBandwidth(uint32_t cents); + + // Gets the gain, in cents, as set. + uint32_t getBandwidth() const { return mBandwidth + 1; } + + // Sets frequency value. Actual change will only take place upon commit(). + // This value will be remembered even if the filter is in disabled() state. + // millihertz Frequency value in mHz. + void setFrequency(uint32_t millihertz); + + // Gets the frequency, in mHz, as set. + uint32_t getFrequency() const { return mNominalFrequency; } + + // Gets gain[dB]/2 points. + // Results in mHz, and are computed based on the nominal values set, not on + // possibly rounded or truncated actual values. + void getBandRange(uint32_t & low, uint32_t & high) const; + + // Applies all parameter changes done to this point in time. + // If the filter is disabled, the new parameters will take place when it is + // enabled again. Does not introduce artifacts, unless immediate is set. + // immediate Whether to apply change abruptly (ignored if filter is + // disabled). + void commit(bool immediate = false); + + // Process a buffer of input data. The input and output should contain + // frameCount * nChannels interlaced samples. Processing can be done + // in-place, by passing the same buffer as both arguments. + // in Input buffer. + // out Output buffer. + // frameCount Number of frames to produce. + void process(const audio_sample_t in[], audio_sample_t out[], + int frameCount) { mBiquad.process(in, out, frameCount); } + + // Enables the filter, so it would start processing input. Does not + // introduce artifacts, unless immediate is set. + // immediate Whether to apply change abruptly. + void enable(bool immediate = false) { mBiquad.enable(immediate); } + + // Disabled (bypasses) the filter. Does not introduce artifacts, unless + // immediate is set. + // immediate Whether to apply change abruptly. + void disable(bool immediate = false) { mBiquad.disable(immediate); } + +private: + // Precision for the mFrequency member. + static const int FREQ_PRECISION_BITS = 26; + // Precision for the mGain member. + static const int GAIN_PRECISION_BITS = 10; + // Precision for the mBandwidth member. + static const int BANDWIDTH_PRECISION_BITS = 10; + + // Nyquist, in mHz. + uint32_t mNiquistFreq; + // Fractional index into the gain dimension of the coef table in + // GAIN_PRECISION_BITS precision. + int32_t mGain; + // Fractional index into the bandwidth dimension of the coef table in + // BANDWIDTH_PRECISION_BITS precision. + uint32_t mBandwidth; + // Fractional index into the frequency dimension of the coef table in + // FREQ_PRECISION_BITS precision. + uint32_t mFrequency; + // Nominal value of frequency, as set. + uint32_t mNominalFrequency; + // 1/Nyquist[mHz], in 42-bit precision (very small). + // Used for scaling the frequency. + uint32_t mFrequencyFactor; + + // A biquad filter, used for the actual processing. + AudioBiquadFilter mBiquad; + // A coefficient interpolator, used for mapping the high level parameters to + // the low-level biquad coefficients. + static AudioCoefInterpolator mCoefInterp; +}; + +} + +#endif // ANDROID_AUDIO_PEAKING_FILTER_H |